[Freeswitch-users] Does FS handle remote NATed phones?
anthmct at yahoo.com
Wed May 23 06:42:15 PDT 2007
We are essentially in BETA right now for the first release so
we would welcome your input as a tester.
FreeSWITCH is more than ready to move SIP traffic, all I would warn is that some of the configuration and behaviors therein have not been finalized and are hence subject to change between now and the final release but there probably aren't many.
Anthony Minessale II
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----- Original Message ----
From: Fred <codecomplete at free.fr>
To: freeswitch-users at lists.freeswitch.org
Sent: Tuesday, May 22, 2007 2:52:12 PM
Subject: [Freeswitch-users] Does FS handle remote NATed phones?
It's been over a week now, and I still can't get voice RX/TX to work
between a Linksys VoIP gateway and a remote IP phone with Asterisk as the PBX:
The phone rings, but no sound either way.
I'm beginning to wonder if Asterisk can handle being behind a NAT, and
bridge calls to remote phones that are themselves behind a NAT :-/ If not,
is FreeSwitch the solution because of its support for STUN? Is it ready for
production, or is it a bit early yet?
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